Session Initiation Protocol (SIP) is rapidly becoming the de facto signaling protocol for establishing, modifying and terminating multimedia sessions between users in a communication network. SIP is described in J. Rosenberg et al., “SIP: Session Initiation Protocol,” Internet Engineering Task Force (IETF) RFC 3261, June 2002, which is incorporated by reference herein. SIP has also been adopted for the IP Multimedia Subsystem (IMS), which is the next-generation core network architecture for mobile and fixed services defined by the 3rd Generation Partnership Project (3GPP).
Further, SIP is commonly utilized in conjunction with Voice over Internet Protocol (VoIP) communications wherein users may participate in voice-based communication sessions (i.e., “calls”) over an IP network. Thus, in this context, SIP is used in call setup, call disconnect and call feature implementation. Other types of multimedia may be transferred over the network in accordance with a SIP-based call.
With more and more deployments of SIP-based networks, carriers have put much more of a focus on providing multimedia services and value-added applications in order to generate new revenue. However, with the addition of multiple applications, the likelihood increases that two or more applications must interact with one another. Unfortunately, the SIP protocol does not adequately support such interworking between multiple application servers running such applications, particularly in accordance with functions such as call transfer.
It is therefore apparent that a need exists for call transfer techniques between multiple application servers, particularly in SIP-based networks.